RE: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Fine but don't mix up Swedish Danish beer ... -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens steve szmidt Verzonden: vr 1/04/2005 16:39 Aan: Asterisk Users Mailing List - Non-Commercial Discussion CC: Onderwerp: Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now On Friday 01 April 2005 02:40, Olle E. Johansson wrote: During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. Olle, you better take a break! For the rest of you, good luck! You'll need it. I think finally the Danish Elephant beer that is so strong has gone to Olle's head. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modify CallerID (on SIP phone) during call
Is it possible to modify the caller id on the phone during a call (session) ? If not does anybody know with which SIP request this could be handled ? I'm know investigating RFC3311 which seems to offer an answer but if somebody already has an answer ... Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma A102 cards testing
We got also these problems and where searching like fools for solutions ... until the time we changed the main board of the server! (Interrupt sharing or Hyper threading stuff, I don't remember) we replaced the Supermicro board with an intel. Try the same config on another machine (maybe an older P3 or P4 or AMD) Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vikram Rangnekar Sent: zondag 13 februari 2005 13:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma A102 cards testing Does anyone have any experience ith configureing the sangoma A102 card for testing using a e1 cross cable i've configured and installed the cards properly even the lights on the card are green which proves that my cross cable is properly built too. my problem is with asterisk which gives me these errors PRI got event: HDLC Abort (6)on Primary D-channel of span 1 PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 No D-channels available! Using Primary on channel anyways 47! PRI: !! Not good - head of queue has not been transmitted yet I've tried everything i can think off with the wancfg configuration files here is my zaptel and zapata configs. span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3 bchan=32-46 dchan=47 bchan=48-62 -- zapata.conf switchtype=euroisdn signalling=pri_net group=1 channel=1-15 channel=17-31 group=2 signalling=pri_cpe channel=32-46 channel=48-62 --- do i need to fool around with some jumpers on the card or something to activate internal clock on the card. zttol says INTERNALLY CLOCKED for both the ports. There are NO Alarms and no missed IRQ's I'm using asterisk 1.0.5 on debian with 2.4.29 kernel -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Dialplan command PPPD released
How are things going for echo and so? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: donderdag 3 februari 2005 21:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dialplan command PPPD released Hi! How's that different from ZapRAS? It doesn't need Zap. :-) BTW, the Sirrix card comes with its own channel driver (currently I have one here to play with). Cheers, Philipp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP600 stuck at Running App = sip.ld(was: Re: Polycom 1.4.1 firmware for IP500/IP600)
Did you try to boot without lan just the power ... I've had this same problem to and rebooted the device without lan connection -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis-David Mitterrand Sent: woensdag 26 januari 2005 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom IP600 stuck at Running App = sip.ld(was: Re: Polycom 1.4.1 firmware for IP500/IP600) On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote: If you have it, can I get a copy please, or possibly can you send it to the keeper of http://www.freedomphones.net/polycom/files/ I am looking for the latest boot image too. 1) I have the 1.4.1 firmware. To whom should I send the files? There is no contact info in this web site. 2) Now I am having a problem with my IP600 test unit: While performing tests on the Polycom IP600 I changed a configuration item and during reboot the phone stopped at the Running App = sip.ld stage and seems stuck there. I reinitialized all configuration files to their defaults from the zip files you sent me, to no avail. Plugging/unplugging the phone does not help as it starts and then stops booting at the same stage, while the message waiting indicator stays solid red (whereas previously it would flash continuously until full startup). Bootrom version: 2.6.1 Sip.ld version: 1.4.1.0040 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600
Does somebody have this new firmware from/for Polycom ? Thanks Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Versatel PRA in Belgium/Netherlands
Problem solved : The reason was quite simple ... but annoying : Interrupts !!! damned !!! Thank you -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Florian Overkamp Verzonden: wo 19/01/2005 10:08 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion' CC: Onderwerp: RE: [Asterisk-Users] Versatel PRA in Belgium/Netherlands Hi, -Original Message- Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume) Yes, we have such setups. Please contact me off-list with some more info about what card you are using etc. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Versatel PRA in Belgium/Netherlands
Versatel uses CRC4 (in Belgium) zaptel.conf : span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 alaw=1-31 -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Florian Overkamp Verzonden: wo 19/01/2005 10:33 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion' CC: Onderwerp: RE: [Asterisk-Users] Versatel PRA in Belgium/Netherlands Hi, -Original Message- Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume) Check with your provider to see if they expect crc on checking or not. I've had to switch it off for serveral installations. Good call: Versatel does not use CRC4 on my link Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Versatel PRA in Belgium/Netherlands
Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soekris net4801 for home use?
And thanks Kristian for the link. I'll make sure to give your distro a shot in case I actually order the box. Just haven't completely made up my mind yet. Nice pics btw you have on your site, what board is that? http://www.pcengines.ch Regards, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on a Linksys WRT54G(S)
Well for example : use SIP on your LAN an use IAX to connect the outside world ... -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Bryan Mannos Verzonden: do 25/11/2004 10:04 Aan: Asterisk Users Mailing List - Non-Commercial Discussion CC: Onderwerp: Re: [Asterisk-Users] Asterisk on a Linksys WRT54G(S) A noble feat to attempt, but I have to ask, why? How on earth would this be a benefit of any real use other than you happen to own one and say you've done it? On Wed, 24 Nov 2004 00:52:29 +0100, Bastian Schern [EMAIL PROTECTED] wrote: Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. winmail.dat___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium E100P or TE410P card
We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100 or TE410 card an PRA line
We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael Sorry for the previous html mail DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 500 software?
http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: zaterdag 6 november 2004 19:48 To: Asterisk-a-users-list Subject: [Asterisk-Users] Polycom 500 software? Just unboxed two brand new SoundPoint IP 500 phones. There was no software shipped with the units. Is the basic sip software (v1.3.1?) available from somewhere this weekend? Tried digging around the Polycom site including registering the product, etc, however there doesn't appear to be any way to download the software necessary to even get the phone operational. Thoughts anyone? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser
They seem to be very good ! but where the hell could we buy them in Europe !? Michael -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens John Todd Verzonden: di 17/08/2004 4:56 Aan: [EMAIL PROTECTED] CC: Onderwerp: RE: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser At 4:16 PM -0700 on 8/16/04, Wiley E. Siler wrote: -Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Monday, August 16, 2004 4:11 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser On Tue, 2004-08-17 at 00:34, Wiley E. Siler wrote: Also, is the new SIP and bootrom release available for download somewhere? Thanks, Wiley Don't know if these are the latest but here are some links. First one has sip bootrom files: http://www.freedomphones.net/polycom/files/ http://www.voip-info.org/wiki-Polycom+Phones Regards, Patrick Yep. Those are the latest. Sip 1.3 and Boot 2.5. Thank you! Wiley Cool; I've been waiting for that release, and I'm very happy to see the XML minibrowser in the SIP image. I dug through the manual, but did not see the feature that I've been waiting for: XML URL fetching based on SIP INVITE or SIP NOTIFY parameters. I'd like to add a SIP header (or partial header) to my INVITEs so that the phone will jump to an XML URL upon receipt of the INVITE. This would allow a screen pop with more useful data in it, instead of just a simple caller ID. I could bring up a picture, a chart, a call history, etc. etc. etc. In other words: a push to the phone for a page based on a realtime event. However, I don't see that in there. Boo hoo. This might be possible with the 'services' button, but that requires user intervention, and probably would have to happen after the phone has been picked up - I haven't tried, and my schedule prevents me from trying at the moment. Anyone have any ideas on this? PS: Everyone who has a Cisco 79xx phone _must_ get a Polycom IP600 somehow on their desk. The administrator's manual has so many cool features that can/should be integrated into Asterisk, that it's worth just reading the PDF to see how useful an IP phone could actually be if we (Asterisk) had support for some of the advanced features like shared line appearances, call bridging, and barge-in. Cisco: listen up. These guys will eat your lunch if you keep crippling the SIP projects. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. winmail.dat
RE: [Asterisk-Users] randomize Dial() target
Use the queue functionality see www.voip-info.org/asterisk -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Marcin Mazurek Verzonden: ma 16/08/2004 15:21 Aan: [EMAIL PROTECTED] CC: Onderwerp: [Asterisk-Users] randomize Dial() target Hi, is it possible to randomize extension which would be choosed by Dial()? I would like to forward phone calls to one of sales rep in randomized way (not to harm anyone;) ). tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. winmail.dat
RE: [Asterisk-Users] CISCO 7960G FIRMWARE
I have to agree that the hardware is fanatastic ... but the functionality is rather poor --- example A --- receive call A answer A Receive call B answer B You can't link/transfer these 2 damned calls. --- example B --- You first have to push new call before dialing a number. this seems to be a detail but 1° why ? and 2° try to migrate users from another system ... --- example C --- an a bit of a proffesional phone transfering a call should be easy and possible in all times : first push more and then transfer again why ??? You only can transfer if you prepared your transfer ! I tend to suspect Cisco not to fine tune their phones for SIP because they are a bit scarry ... for their callmanager ? Give me a combination of the HArdware of cisco and the software of a Snom200 and THAT would be a GREAT phone ! Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Shaun Ewing Sent: donderdag 15 juli 2004 8:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CISCO 7960G FIRMWARE Nonsense. If you have access to the firmware, they're fantastic phones and the best phones I've ever used. DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CISCO 7960G FIRMWARE
like i said this is a detail look at the other comments but to clearify my point : Try to dial a number without pushing a button or picking up the handset. (dial your number and then pickup the handset or indeed with a dialplan it uses automatically the speakerphone) This is so easy to make it work but why does Cisco not do it ?? but i use also the cisco 7905G and their every point i made against the 7960 does work !? but it is not handsfree !!! it sounds strange but i prefer the 7905g Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Shaun Ewing Sent: donderdag 15 juli 2004 11:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CISCO 7960G FIRMWARE On Thu, 15 Jul 2004 10:08:41 +0200, Michael Devenijn [EMAIL PROTECTED] wrote: --- example B --- You first have to push new call before dialing a number. this seems to be a detail but 1° why ? and 2° try to migrate users from another system ... I don't know how your handsets are setup, but I have *never* had to do that with my 7940/7960 phones (unless using speakerphone, in which case it's either New Call or the speaker button). I simply pick up the phone and dial away. Because I have a dialplan setup, the phones sense when enough digits have been dialed and sends the call immediately. All people need to remember here is to dial 0 before the external number. My 80 year old Grandmother has even used the Cisco phones with no problems, and she hasn't even used a computer before. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiSIP and Zyxel Prestige 2000W
It seems we have a more recent version (but i do not remember where i got it, i think it is frome a post on this list) wj000e_img.ftp in wj000e_img.zip Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dominique Kull Sent: woensdag 14 juli 2004 9:41 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] WiSIP and Zyxel Prestige 2000W I can confirm that WEP with Netgear's ME103 is no problem. Latest firmware I found was here: http://www.zyxel.co.uk/support/ukadslfw.php DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS on TE410P
In belgium people can receive SMS's on fixed lines, my question is if this feature is possible on the digium TE410P card in combination with asterisk ? Kind regards Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Disappointed
Yes it is possible, with the chan_CAPI drivers from junghanns.net i only used the 4BRI cards from Eicon but they are similar to the PRI cards i didn't have any ISDN knowledge before. but first tried to install the card with CAPI on a redhat 9 machine with exactly the description from eicon then tried to start the chan_capi and it finally worked !! It took me some research as above all i was not really a linux guru and about the conferencing question = yes Michael -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Calum Verzonden: ma 28/06/2004 11:11 Aan: [EMAIL PROTECTED] CC: Onderwerp: [Asterisk-Users] Disappointed Well, I have to confess that I am disappointed that in a fairly high volume list like this, I haven't had one reply to the questions I've asked. (I know I haven't got any right to expect a reply, but communities are usually fairly helpful). It might be really obvious to you guys, but if you have not a lot of experience with ISDN/PBXs, it's hard to understand. I'm going to unsubscribe, so if anyone feels that they can help me out, please reply to my email address. ( calum dot asterisk **at** umtstrial dot c o dot u k ) Does this card work/can it be made to work with Asterisk? lspci: 07:06.0 Network controller: Eicon Technology Corporation DIVA Server PRI-30M 2.0 Can I establish 2 outbound calls with it, and conference them together? Thanks once again. Don't bother with flames. Calum -- Random russian saying: If the thunder is not loud, the peasant forgets to cross himself. jabber: [EMAIL PROTECTED] pgp: http://gk.umtstrial.co.uk/~calum/keys.php Linux 2.6.5-gentoo 10:06:14 up 19 days, 22:34, 1 user, load average: 0.35, 0.31, 0.29 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. winmail.dat
RE: [Asterisk-Users] Video/H323/SIP
I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based solution to establish a video connection with * and sip protocol? - Original Message - Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Generator for ISDN (PRI/BRI)
why not use asterisk with QaudBRI and/or E100P ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miroslav Nachev Sent: woensdag 23 juni 2004 16:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Generator for ISDN (PRI/BRI) Hi, I am looking for Call Generator for PRI ISDN and BRI ISDN signals. From where I can found some cheap or 2nd hand call generator (tester/analyzer)? Maybe PCI based will be cheaper than standalone solution. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P / Eicon PRI
for today we only have experience with BRI applications together with asterisk. is the following scenario possible and stable enough for production? FYI : We want to build a unified messaging application integrated with SIP. We have an E1 connection in Belgium with 100 msn's We would think about having 2 servers : Server A : Asterisk PRI card (Digium TE410P) Server B : Fax server PRI card (Eicon PRI30M) Call --- TE410P/1 --- Asterisk Extension --- Voice ? --- Voicemail or Dial Fax ?--- TE410P/2 crossover to --- Server B (Eicon PRI) Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P / Eicon PRI
i'dd like to but is it stable enough for production (receiving over 500 faxes a day ?) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter Junghanns Sent: vrijdag 18 juni 2004 13:58 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TE410P / Eicon PRI We would think about having 2 servers : Server A : Asterisk PRI card (Digium TE410P) Server B : Fax server PRI card (Eicon PRI30M) Call --- TE410P/1 --- Asterisk Extension --- Voice ? --- Voicemail or Dial Fax ?--- TE410P/2 crossover to --- Server B (Eicon PRI) save 10k EUR and use spandDSP (www.opencall.org) for fax instead of the second server with the Eicon PRI card. Michael best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Siemens Optipoint 400 standard SIP
Dear, Is there somebody who have experience with the Siemens Optipoint 400 standard SIP ? And where can we buy it (i'm from belgium) We are using for the moment Cisco 7960, 7905, snom 200, Mitel 5055 and in my opinion the Mitel does his work the best combined with the 7905, the 7960 is realy anoying to transfer, On the snom 200 you can't dial a number veryfast due to the rubber buttons Thank you Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem compiling zaptel with BRIstuff 0.0.2
This is what i got ... and i could not find the problem by my own /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/qozap.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/tor2.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wcfxs.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wct1xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wct4xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/ztd-eth.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/ztdynamic.o [ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf [EMAIL PROTECTED] zaptel_bri]# DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem compiling zaptel with BRIstuff 0.0.2
yes i installed it ... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Frederic Olivie Sent: woensdag 19 mei 2004 12:44 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem compiling zaptel with BRIstuff 0.0.2 Have you installed the zaptel module (zaptel.o) in your modules ? Try an : insmod zaptel If it does not work, it means it has not been installed. - Original Message - From: Michael Devenijn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, May 19, 2004 10:45 AM Subject: [Asterisk-Users] Problem compiling zaptel with BRIstuff 0.0.2 This is what i got ... and i could not find the problem by my own /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/qozap.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/tor2.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wcfxs.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wct1xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wct4xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/ztd-eth.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-1.2115.nptl/misc/ztdynamic.o [ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf [EMAIL PROTECTED] zaptel_bri]# DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with QuadBRI
- I'm not a Linux user i'm trying to get in it ... - Fedora core 1 - QuadBRI card bought from junghanns.net - We want to use the card in TE mode to connect to the TELCO - Downloaded BRISTUFF0.0.2(stable) latest from junghanss.net/asterisk - followed the instructions on voip-info.org - compiled everything still got the error when make load in qozap : insmod ./qozap.o ./qozap.o: unresolved symbol zt_ec_chunk ./qozap.o: unresolved symbol zt_unregister ./qozap.o: unresolved symbol zt_transmit ./qozap.o: unresolved symbol zt_receive ./qozap.o: unresolved symbol zt_register make: *** [load] Error 1 when i execute ztcfg -v -c /etc/zaptel.conf we get : Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) 12 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) Other info : When i look in the BIOS i see the card is on IRQ5 alone When i use cat /proc/interrupts i see nothing on IRQ5 zaptel.conf : ... loadzone=nl defaultzone=nl #fxsks=1 span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 200
Very good phone especially with the latest firmware ... Major point : the soft rubber keys are to hard to push, you can't make a number fast enough Minor : the phone is too light Good : - Multi national configurations / tones etc... - Good design - ... the others told it already Michael Devenijn Telessence -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hermann Wecke Sent: woensdag 12 mei 2004 19:33 To: Asterisk Mailling List Subject: [Asterisk-Users] SNOM 200 Sorry to ask this here but I believe that it is the best place to receive a feedback... I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *, and the overall impression about these phones... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Vegastream 50 BRI
Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... sip.conf extract : [gw001] type=friend host=dynamic defaultip=192.168.0.12 nat=no dtmfmode=rfc2833 canreinvite=yes qualify=no context=tlsgw extensions.conf extract (from the contact [tlsgw]) : exten = 57228047,Dial(SIP/cs001,40,tr) ... Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 sip:[EMAIL PROTECTED];tag=-0002-3A81BAD9 To: sip:[EMAIL PROTECTED] Max-Forwards: 70 Call-ID: [EMAIL PROTECTED] CSeq: 83606 INVITE Contact: sip:[EMAIL PROTECTED]:5060;maddr=192.168.0.12 Supported: replaces Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER Accept-Language: en Content-Type: application/sdp Remote-Party-ID: 478758923 sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off Content-Length: 178 v=0 o=Vega50 3 1 IN IP4 192.168.0.12 s=Sip Call t=0 0 m=audio 10004 RTP/AVP 8 0 18 c=IN IP4 192.168.0.12 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 14 headers, 9 lines Using latest request as basis request Sending to 192.168.0.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMA Found description format PCMU Found description format G729 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 sip:[EMAIL PROTECTED];tag=-0002-3A81BAD9 To: sip:[EMAIL PROTECTED];tag=as1fa83a23 Call-ID: [EMAIL PROTECTED] CSeq: 83606 INVITE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.12:5060 dkmapbx*CLI Sip read: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 sip:[EMAIL PROTECTED];tag=-0002-3A81BAD9 To: sip:[EMAIL PROTECTED] Max-Forwards: 70 Call-ID: [EMAIL PROTECTED] CSeq: 83606 ACK Contact: sip:[EMAIL PROTECTED]:5060;maddr=192.168.0.12 Content-Length: 0 9 headers, 0 lines DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need an example of using the directory command
Go on www.voip-info.org an search for IVR examples ... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Mahler Sent: Saturday, March 20, 2004 5:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need an example of using the directory command Does someone please have a sample that shows how to use the directory command in extensions.conf? Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with Vegastream 50 BRI
sorry, the sip extract is from a previous test now i get the same problem but with looking for 57228047 in tlsgw and it's the same error, it searching in this direction : why are the 2 ast values 0 ?? Non-codec capabilities: us - 1, them - 0, combined - 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Saturday, March 20, 2004 11:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem with Vegastream 50 BRI Michael Devenijn wrote: Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... sip.conf extract : [gw001] type=friend host=dynamic defaultip=192.168.0.12 nat=no dtmfmode=rfc2833 canreinvite=yes qualify=no context=tlsgw extensions.conf extract (from the contact [tlsgw]) : exten = 57228047,Dial(SIP/cs001,40,tr) ... Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Asterisk isn't looking in the context tlsgw, for some reason it checks in the sip context. If this is your default context, Asterisk doesn't connect the incoming call with gw001. You have host=dynamic - is the gateway registred with Asterisk at all? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPC5000 (WIP-5000 from hitachi cable)
My review for this wireless SIP phone Got it friday tested it this weekend with asterisk ... everything went well (could just place one call) and then i've got authentication faults but i think it's a asterisk isue that could be solved ... i've changed the authentication settings on asterisk to accept anything and everything went well (simply placing a call, reciving a call) But saturday (after charging the batteries), nothing ! I just could start it up but then the software seems to be blocked completely and i couldn't find any software reset or upgrade ... Some major issues : - couldn't see on the screen when a call was missed - couldn't transfer a call (except with the asterisk #) - phone loses registration, need to push the reg key after 10 min) (here stops my test with the software side of the phone due to the lock) - some buttons (8 and down) don't work properly ... - it is a early test unit, this phone is not really for the market see software version (0.0.2) which says enough about the buggy state of the software. but if they are working on the software (a lot) it would be a great phone very light weight, very clear screen , good useability Kind regards Michael Devenijn -Oorspronkelijk bericht- Van: Craig Waddington [mailto:[EMAIL PROTECTED] Verzonden: ma 15/03/2004 11:07 Aan: Michael Devenijn CC: Onderwerp: FW: IPC5000 FYI _ From: FahdTel AB [mailto:[EMAIL PROTECTED] Sent: 12 March 2004 17:47 To: Craig Waddington Subject: RE: IPC5000 Hello Craig, Thank you for your kind inquiry. Pls find attached handset and pricing information. Best Regards Mohammed -Original Message- From: Craig Waddington [mailto:[EMAIL PROTECTED] Sent: den 11 mars 2004 19:04 To: [EMAIL PROTECTED] Subject: IPC5000 Hi, I am looking to purchase some of these phones. Can you provide me with information and prices please. Thank you, Craig Waddington. DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UDC SYSTEMS
Does anybody have experience with these units ?? http://www.udcsystems.com/ DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPC5000 - Wireless Sip phone
no i bought this one -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Craig WaddingtonSent: Thursday, March 11, 2004 8:58 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] IPC5000 - Wireless Sip phone Thanks for the info. Sounds good. Does that mean I can contact them for a test unit also, to try before I buy? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael DevenijnSent: 11 March 2004 18:25To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] IPC5000 - Wireless Sip phone I ordered a test unit and will recieve it this week (already shipped from sweden), i will post some comments on this list when it is tested .. I hope it will do his job !! ... the mail they sent to : Hello Michael, Hope you are well. Your sample is on the way and pls find attached delivery note for your reference. Ps. frieght charge was USD10 lower, so we own you USD10 that we will pretty reduced it with your next order or we transfer it to your bank account. I'll the coming days send you updated information about the handset and its new design i.e. it has L2 roaming feature now. The handoff time is 200 ~ 300ms between the AP. We aim to short it to 100 ~ 200ms. The implementation of Web Authentication(web-login) what we call HTTPS(SSL)is ongoing and should be releasedon June. It can be software upgrade. Best Regards,Mohammed Fahd -Oorspronkelijk bericht- Van: [EMAIL PROTECTED]namensCraig Waddington Verzonden: do 11/03/2004 19:15 Aan: [EMAIL PROTECTED] CC: Onderwerp: [Asterisk-Users] IPC5000 - Wireless Sip phone I am looking to buy a wireless sip phone, probably the IPC5000, I have looked at Wisip phone and read tons of posts regarding that phone. Do any * admins have any feedback on this phone? Is there any major differences between the phones, besides looks? The site has very limited information regarding prices etc. Ta. DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer.
RE: [Asterisk-Users] windows alternitives to Asterisk?
I'm working on a abstraction toolset for asterisk, some screenshots : http://www.dkma.be/devenijn/data/tls.jpg http://www.dkma.be/devenijn/data/tls1.jpg http://www.dkma.be/devenijn/data/tls2.jpg http://www.dkma.be/devenijn/data/tls3.jpg http://www.dkma.be/devenijn/data/tls4.jpg some technical details : - Fully based on a Postgresql DB (+/- 40 tables) - Extensive use of macro's in the extensions.conf - some little patches to the asterisk source (additions) to extend the manager capability - a layer between the manager port and the PGSQL written in C (to prevent that each separate client would connect to the manager interface) - some AGI scripts (written in C) to fulfill some needs which are not build in asterisk for example : redirect through dragdrop. - A program (written in C) which extracts all the info from de Pgsql and converts it to the needed * conf files upon request - goal : everything works on Win32 clients : configuration, outlook integration , panel programs, attendant program ... so we could sell a standard box plug it to the PSTN network and the ethernet, install the sotware on a W32 client and off they go without the explicit knowledge of linux. the sound files (will be uploaded on a FTP on the box and copied to the appropriate destination with the some program as one point above I especially developed these toolset arround asterisk, so i do not need the take in account the Asterisk license except for the patches which will be published. development status : - First tests are positive - Working on the last big point is the extended (graphical call router to easily design IVR's) - Working on a .NET SDK which also is a abstraction. - Timing end of march ... but as everybody knows it could take some more time. Some personal statements : - I know it is not very good to develop for windows but this is the only way for me and for th major Small businesses outside ... and if there are some enthousiast to convert it to a linux platform ... go ahead. - For the moment it is quit oriented on a combination of ISDN hardware (capi.conf) and cisco phones. once this is working without major bugs i will extend it to (IAX.conf ... and other things) - My only goal to develop this is to get asterisk on bigger user basis - What i regret is the lack of developement of a interface between asterisk and other software (for windows TAPI for example) i asked many times some basic questions about the manager interface but nobody seems to be interested ! i strongly belief this would be a major break through. but i do not have enough C++ and/or Linux knowledge (and TIME ) to develop these features. The sourcecode ??? - Well lets play it tricky ... i'm going to publish all of the code under GPL if i see some changes to latest point i mentioned just above. (and don't try to convince me with ethical arguments, i'm now trying for 3 years to make some descent money with integration of open source software and i see it as a win/win operation) Just fire me !!! -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Tilghman Lesher Verzonden: ma 8/03/2004 21:46 Aan: [EMAIL PROTECTED] CC: Onderwerp: Re: [Asterisk-Users] windows alternitives to Asterisk? On Monday 08 March 2004 13:59, hank smith wrote: is there a program that I can install on my linux box so I can configure the pbx from the internet from my windows box so I don't have to work with config files? In a word, no. There are a few GUI applications in the process of being developed, but most of them are a substitute only for using an editor on the Linux machine, not a full-fledged abstraction from the configuration. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. winmail.dat
RE: [Asterisk-Users] windows alternitives to Asterisk?
- The whole toolset running on W32 is already ODBC (in fact it's the only way ... ) - Once i reach stabilty i will try to convert it to fully DB independant - The reloading thing ... i leave it to others ... once it is possible i'll make things executable -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Michael Shuler Verzonden: di 9/03/2004 9:40 Aan: [EMAIL PROTECTED] CC: Onderwerp: RE: [Asterisk-Users] windows alternitives to Asterisk? Contact me at your earliest convenience, we would be immediately interested in the product for our CLEC for provisioning services to our customers. Couple of issues for us: - Would need to support MySQL and Oracle (should be easy to switch to ODBC instead of Postgresql specific functions) - Need a way to change settings on * without reloading... I found something that is no longer maintained that can do this, see http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DynExtenDB We already got the SIP features in http://bugs.digium.com/bug_view_page.php?bug_id=0001086 applied which takes care of adding customers - Would need to support virtual PBX concept for a large carrier deployment and thousands of businesses - Would need to support management of clustered * servers which should be taken care of by the above 2 links and a shared file system for VM and ACD (which would be cool if it was stored in an SQL DB too) And much much more but the above a biggies for us. Michael Shuler, C.E.O. BitWise Systems, Inc. 1301 W. Pioneer Parkway Peoria, IL 61615 Office: (217) 585-0357 Cell: (309) 657-6365 Fax: (309) 213-3500 E-Mail: [EMAIL PROTECTED] Customer Service: (877) 976-0711 -Original Message- From: Michael Devenijn [mailto:[EMAIL PROTECTED] On Behalf Of Michael Devenijn Sent: Tuesday, March 09, 2004 2:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] windows alternitives to Asterisk? I'm working on a abstraction toolset for asterisk, some screenshots : http://www.dkma.be/devenijn/data/tls.jpg http://www.dkma.be/devenijn/data/tls1.jpg http://www.dkma.be/devenijn/data/tls2.jpg http://www.dkma.be/devenijn/data/tls3.jpg http://www.dkma.be/devenijn/data/tls4.jpg some technical details : - Fully based on a Postgresql DB (+/- 40 tables) - Extensive use of macro's in the extensions.conf - some little patches to the asterisk source (additions) to extend the manager capability - a layer between the manager port and the PGSQL written in C (to prevent that each separate client would connect to the manager interface) - some AGI scripts (written in C) to fulfill some needs which are not build in asterisk for example : redirect through dragdrop. - A program (written in C) which extracts all the info from de Pgsql and converts it to the needed * conf files upon request - goal : everything works on Win32 clients : configuration, outlook integration , panel programs, attendant program ... so we could sell a standard box plug it to the PSTN network and the ethernet, install the sotware on a W32 client and off they go without the explicit knowledge of linux. the sound files (will be uploaded on a FTP on the box and copied to the appropriate destination with the some program as one point above I especially developed these toolset arround asterisk, so i do not need the take in account the Asterisk license except for the patches which will be published. development status : - First tests are positive - Working on the last big point is the extended (graphical call router to easily design IVR's) - Working on a .NET SDK which also is a abstraction. - Timing end of march ... but as everybody knows it could take some more time. Some personal statements : - I know it is not very good to develop for windows but this is the only way for me and for th major Small businesses outside ... and if there are some enthousiast to convert it to a linux platform ... go ahead. - For the moment it is quit oriented on a combination of ISDN hardware (capi.conf) and cisco phones. once this is working without major bugs i will extend it to (IAX.conf ... and other things
RE: [Asterisk-Users] Hylafax integration
I'm a bit of a linux zero (at least i'm working to know it but it takes some time)
Do i understand :
The call comes in on a card (in this case zaptel) for exampl,e and if in the dialplan
you have an 'f' extension Hylafax will get the communication from the zaptel card ? or
is it simply a redirect to another extension or tel. number ?
thank you
Michael
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Darren Nickerson
Verzonden: ma 8/03/2004 15:08
Aan: [EMAIL PROTECTED]
CC:
Onderwerp: Re: [Asterisk-Users] Hylafax integration
Alessio,
What seems to be the problem? In principle, it's the same as if you had a
conventional fax machine. As long as you have a fax extension ('f') in your
dialplan, inbound calls should be routed properly and HylaFAX (if there's a
faxgetty running on the modem) should pick up and receive the fax.
-Darren
--
Darren Nickerson
Senior Sales Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 ext 8106 office
+1.215.243.8335 fax
- Original Message -
From: Alessio Focardi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 08, 2004 8:51 AM
Subject: [Asterisk-Users] Hylafax integration
Hi,
I have a working install of asterisk that I would like to integrate
with Hylafax: an asterisk extention must be transfered to Hyla for fax
receiving.
I have roamed around in google but there is very few information on
that matter.
Anyone can help me ?
p.s.
Hyla is already up and running in my pc.
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain information which is
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[Asterisk-Users] Snom phones
Did somebody already get a snom 220 phone ?? DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T.38 fax (off-topic)
Does somebody now if there is some opensource software which can handle T.38 SIP and convert it to a tiff or something ? Kind regards Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vegastream 50 BRI
does sombody has an example config for a vegastrem 50 BRI and asterisk would help me a lott thanks Michael DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk)
Jan, Where can we get any technical documentation about sccp protocol i've searched with google and at cisco but i don't find anything useful ... Michael Van: Jan Czmok [mailto:[EMAIL PROTECTED] Verzonden: wo 21/01/2004 15:04 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] I got it (was: Cisco 7940 with asterisk) Jeff Gustafson ([EMAIL PROTECTED]) wrote: Hi again, I found chan_skinny and that seems to work pretty good. the SCCP one filled out all the buttons really nice, but skinny seems to be working. How do I fill out the second line button on the phone with skinny.conf? Thanks much! Define a second section, however, you also might want to take a peek at chan_sccp. We are currently reworking the complete chan_sccp to support all functions known by the Skinny Protocol. Yes, EVEN the 7920 is working with it now :-) --jan -- Jan Czmok, Network Engineering Support, Global Access Telecomm, Inc. Ph.: +49 69 299896-35 - fax: +49 69 299896-40 - sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] * crashed
May be it's due to a kernel patch ... ?? Try to recompile zaptel, asterisk, ... Van: Jess Magnaye [mailto:[EMAIL PROTECTED] Verzonden: wo 7/01/2004 17:36 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] * crashed I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this error when im trying to connect, and then it suddenly dies out. ERROR[1074412224]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk I tried to run it again using asterisk -gc and I got the ff error: WARNING[1074412224]: File loader.c, Line 312 (ast_load_resource): chan_zap.so: load_module failed, returning -1 Segmentation fault (core dumped) Not sure why. It looks to me it got corrupted after my reboot during change of IP. (Can someone shed light on this?) Thanks. winmail.dat
RE: [Asterisk-Users] AGI Scripting
Use the callmanager their u can use the link event Michael Van: Luciano Ramos [mailto:[EMAIL PROTECTED] Verzonden: di 6/01/2004 13:56 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] AGI Scripting Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano winmail.dat
RE: [Asterisk-Users] Got SIP response 482 Loop Detected
Code you gave the audio settings of your 7905 and which type and firmware for the snom Van: tony banks [mailto:[EMAIL PROTECTED] Verzonden: di 6/01/2004 18:23 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] Got SIP response 482 Loop Detected Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this problem was gone, without me doing anything..Has anyone observed this thing before... Called 810 -- SIP/810-b6dc is ringing -- SIP/810-b6dc answered SIP/910-6c4e -- Attempting native bridge of SIP/910-6c4e and SIP/810-b6dc WARNING[1227879616]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 = 524302 is not codec1 = 524302, can't do reinvite -- Got SIP response 482 Loop Detected back from 129.82.44.226 WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Response) WARNING[1142106560]: File chan_sip.c, Line 2329 (__transmit_response): Unable to determine sequence number from '' Thanks Tony http://clients.rediff.com/signature/track_sig.asp winmail.dat
RE: [Asterisk-Users] AGI Scripting
sorry i mean the manager (on port 5038) see the wiki http://www.voip-info.org/wiki-Asterisk+manager+api Michael Van: [EMAIL PROTECTED] namens Luciano Ramos Verzonden: di 6/01/2004 20:12 Aan: [EMAIL PROTECTED] Onderwerp: RE: [Asterisk-Users] AGI Scripting Which Call Manager??? Luciano -Mensaje original- De: Michael Devenijn [mailto:[EMAIL PROTECTED] nombre de Michael Devenijn Enviado el: Martes 6 de Enero del 2004 10:10 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] AGI Scripting Use the callmanager their u can use the link event Michael Van: Luciano Ramos [mailto:[EMAIL PROTECTED] Verzonden: di 6/01/2004 13:56 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] AGI Scripting Hi!. Is there any way to know which extension answered a call , when dialing from an AGI Script?? Thanks! Luciano winmail.dat
FW: [Asterisk-Users] SIP to SIP redirect while ringing
I didn't get any response on that question, so i supose this feature is possible but there isn't an implementation of it. I'm ready to sponsor this feature in the manager interface (i tried the redirect command but it doesn't work) can somebody help me ?? this feature would make it possbile to use drag drop features ... Kind Regards Michael Devenijn Van: Michael Devenijn Verzonden: ma 24/11/2003 14:38 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] SIP to SIP redirect while ringing is it possible to transfer a call while it's ringing ?? SIP/cs001 calls SIP/cs002 The SIP/cs002 user transfers the call to SIP/cs003, where on SIP/cs003 the phone continues to ring ... inone way or another (trough manager API or something else, don't care) i tried redirect with the manager but it doesn't work (or i didn't understand it) Thank you for any help Michael
[Asterisk-Users] License questioni supose ??
I have some strange question bout the asterisk (gpl license ...) but i'm not an experienced linux user ... What happens if for example a big company buys digium , do we have a garantuee that asterisk stays opensource ??? Kind regards Michael Devenijn winmail.dat
RE: [Asterisk-Users] Asterisk + CRM
Which events did you add ? Van: Jonathan Tew [mailto:[EMAIL PROTECTED] Verzonden: di 23/12/2003 16:25 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Asterisk + CRM We're starting to integrate * with our customer service software. Basically we're pulling off events from the management interface. We're also making some small patches to the code to deliver more events about the channel variables, etc. Anton Yurchenko wrote: Hello, Anyone aware of any CRM products projects that intagrete with *? Or that integrate with any telephony products? Is there some open API for such integration, or are they all proprietory? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
[Asterisk-Users] Manager API Problem
Everythings works great with asterisk exept one feature with redirect : it doesn't redirect when ringing ... BTW are their any plans to extend the manager API ?? Michael Devenijn
RE: [Asterisk-Users] SIP (peer to peer?)
You're right for pure SIP configurations , but Asterisk acts here as media gateway and treats all of the media comm. e.g. to eventually communicate with other types (like PSTN, H323, AIX, ... the voicemail app.) Michael Devenijn IT Manager DKMA Schaarbeeklei 636 B-1800 Vilvoorde Tel.: +32 2 255 10 19 Fax: +32 2 251 03 12 Van: Wim Venneman [mailto:[EMAIL PROTECTED] Verzonden: ma 8/12/2003 22:17 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] SIP (peer to peer?) Hi all, Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer) Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? Maybe a stupid question, but I'm not a SIP expert. Thank you for your help. Wim winmail.dat
RE: [Asterisk-Users] CTI/TAPI
I'm a windows developer ready to develop TSP/MSP (TAPI) drivers for asterisk PBX but i searched a little more support to develop these drivers ... unfortunatly i have to develop the drivers commercially because i will need to hire a asterisk freak to explain me in detail how everything works on the side of asterisk so i can implement in the most secure and correct way a TAPI driver on a windows platform for asterisk running on linux (which will for example do his communication with the manager API). but i think we will need to do some additional development of the manager API (which will stay free and eventually communnicated to those who are interested to be in accordance to the asterisk license) but the windows developement will stay our propiarity (simply because we are not an big budget company and we want to get our children some food at the end of the month) Just to inform the community ... i received an offer last week for 1 week of asterisk training +/-2USD !! We can't aford this ! Kind regards Michael Devenijn DKMA Van: Harald Baron [mailto:[EMAIL PROTECTED] Verzonden: di 2/12/2003 9:38 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] CTI/TAPI Hi I want to connect a Windows machine over TAPI with the Asterisk PBX. So is it possible to connect the Windows machine directly to Asterisk (Zaptel card)? Thanks Harry Baron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] CTI/TAPI
You hear it very well ! As i think i'm polite, so i'm not going to put the name of the company online but believe me their was a hole in the seiling after i read the email ... Van: Roy Sigurd Karlsbakk [mailto:[EMAIL PROTECTED] Verzonden: di 2/12/2003 15:23 Aan: Asterisk Users Onderwerp: RE: [Asterisk-Users] CTI/TAPI Just to inform the community ... i received an offer last week for 1 week of asterisk training +/-2USD !! We can't aford this ! Is that USD 20.000,- as in twenty thousand US dollars, or is have someone played around with the keyboard? If so - who the fuck can afford to pay such a price for a week's training??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] CTI/TAPI
the details : 16000USD for the training + 10% for administration + Travel costs to Belgium (from canada) + Hotel costs ... so 'im not far from 2USD ...( it could be even more ) We have a budget but not such a big one ... and even if i would, i would be a bad manager to accept such a cost considering to having a high level engineer (very high) on my pay rol costs something like 8000? a month all in ... so calculate yourself ... Van: Scott Stingel [mailto:[EMAIL PROTECTED] Verzonden: di 2/12/2003 16:18 Aan: [EMAIL PROTECTED] Onderwerp: RE: [Asterisk-Users] CTI/TAPI Must have included a week in Amsterdam Scott M. Stingel Emerging Voice Technology Inc. URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Tuesday, December 02, 2003 2:23 PM To: Asterisk Users Subject: RE: [Asterisk-Users] CTI/TAPI Just to inform the community ... i received an offer last week for 1 week of asterisk training +/-2USD !! We can't aford this ! Is that USD 20.000,- as in twenty thousand US dollars, or is have someone played around with the keyboard? If so - who the fuck can afford to pay such a price for a week's training??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] CTI/TAPI
We are a little Belgian company working for the automotive world (car builders, leasing companies , ...) and believe me if i make an offer of 2USD for a week i'll never have to make an offer again ! I also worked in the past (2000 which is considered as the craziest year for Tech companies) with consultants from one of the big consultant companies worldwide (PW) and a TOP TOP consultant never reached this rates ... Van: mattf [mailto:[EMAIL PROTECTED] Verzonden: di 2/12/2003 16:59 Aan: '[EMAIL PROTECTED]' Onderwerp: RE: [Asterisk-Users] CTI/TAPI maybe it means United States Dimes :) $2,000 ain't bad for a week of training. But to answer your question, I have a friend that does Checkpoint firewall training/consultation and he gets upto $20,000 per week for running training classes. Not in the US mind you but abroad, mostly in Europe. He says American companies are too cheap. MATT--- -Original Message- From: Michael Devenijn [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 9:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] CTI/TAPI You hear it very well ! As i think i'm polite, so i'm not going to put the name of the company online but believe me their was a hole in the seiling after i read the email ... Van:Roy Sigurd Karlsbakk [mailto:[EMAIL PROTECTED] Verzonden:di 2/12/2003 15:23 Aan:Asterisk Users Onderwerp:RE: [Asterisk-Users] CTI/TAPI Just to inform the community ... i received an offer last week for 1 week of asterisk training +/-2USD !! We can't aford this ! Is that USD 20.000,- as in twenty thousand US dollars, or is have someone played around with the keyboard? If so - who the fuck can afford to pay such a price for a week's training??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] * Party in Paris
count me in Michael devenijn DKMA Van: Mark Spencer [mailto:[EMAIL PROTECTED] Verzonden: zo 30/11/2003 6:28 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] * Party in Paris I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] files for upgrade cisco 7960 phone
You have to buy a Cisco contract so you can download the files on their site, but here is a link with the explanations, because once you have the good firmware ... there is a way to go : http://www.loligo.com/asterisk/cisco/79xx/ Michael Devenijn DKMA Schaarbeeklei 636 1800 Vilvoorde Tel: +32 2 255 10 19 Fax: +32 2 251 03 12 Van: Carlos Valdes [mailto:[EMAIL PROTECTED] Verzonden: vr 28/11/2003 7:38 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] files for upgrade cisco 7960 phone hi, some on can send me the files for upgrade cisco 7960 phone now is at P0S30202 or where can download ??? thx [EMAIL PROTECTED] winmail.dat
RE: [Asterisk-Users] RFC3389 support incomplete
Just turn off the silence suppresion Van: Jorge Cisneros Flores [mailto:[EMAIL PROTECTED] Verzonden: vr 28/11/2003 5:57 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] RFC3389 support incomplete Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Who i turn off and how i fix this thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
RE: [Asterisk-Users] Multi-line TTS Outbound Dialer
We are working with realspeak and it is a wonderfull product (even in product) it supports up to 20 languages and has aquired a really good prod. stability ! Van: Steve Underwood [mailto:[EMAIL PROTECTED] Verzonden: vr 28/11/2003 4:41 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Multi-line TTS Outbound Dialer Carl Youngblood wrote: What is EAGI? I will probably use festival for the time being, but I thing that I would eventually like to use ScanSoft's RealSpeak SDK because it is so life-like. Unfortunately our text alerts are fully customizeable, so we can't pre-record them. Beware the likelike TTS, that sucks up thousands of dollars and gets thrown away. RealSpeak is great for demos :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
[Asterisk-Users] Asterisk Training
Dear All, We are all Windoze C++ developers but are working on C/C++ development for linux. But we needsomeone of our teamto be trained quite soon in Asterisk development to add some features we really need inside the company. We are searching somebody or a company who will train a developer during a week (i estimate) We are from Belgium ... so if there is a experienced Asterisk developer in Europe it would help I sent an email to digum sales but didn't get any response. and it is quite urgent We are ready to pay the price which is needed and to travel if necesairy, but we need a person who knows the Asterisk sources very well ! Please can somebody make use a good offer for a training of 1 week in Asterisk development, it is quite logic that this will result in Open source contributions ... and wil bring some extracommercial experience to Asterisk. kind regards Michael Devenijn DKMA bvba Schaarbeeklei 636 1800 Vilvoorde Tel.: +32 2 255 10 19 Fax : +32 2 251 03 12
[Asterisk-Users] SIP to SIP redirect while ringing
is it possible to transfer a call while it's ringing ?? SIP/cs001 calls SIP/cs002 The SIP/cs002 user transfers the call to SIP/cs003, where on SIP/cs003 the phone continues to ring ... inone way or another (trough manager API or something else, don't care) i tried redirect with the manager but it doesn't work (or i didn't understand it) Thank you for any help Michael
RE: [Asterisk-Users] Which ISDM BRI Card for Asterisk?
I'm testing the Eicon Server BRI/PRI cards for the moment and they are very satisfactory because they have some interesting features onboard like echo cancelation () and onboard encoding see www.eicon.com Van: Daniel ANDRE [mailto:[EMAIL PROTECTED] Verzonden: vr 21/11/2003 9:48 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] Which ISDM BRI Card for Asterisk? Hello all, I wonder to have some feedback on using ISDN BRI Cards with Asterisk and the Echo problem. I have tried a simple BRI card with i4l driver and encounter huge echo problem. I have tried to solve it with a Sw chocanceller without success. What I'd like to know is wether some of you have used other BRI Cards (I have seen reference to Eicon cards on this list) and if the echo disappeares with these cards? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
[Asterisk-Users] DIGI Datafire QuadMicro
Did anybody tried this card with asterisk ? http://www.digi.com/pdf/prd_mca_datafirequad.pdf
[Asterisk-Users] TAPI development
Has anyone ever worked opn TAPI stuff to make asterisk work with it ? I'm a Windoze C++ developer dig'n into asterisk (and linux at the same time)since a few months and i'm quite interested in creating a TAPI driver for asterisk. so if anybody did any research in that way please inform me. Also i've you think it's quite impossible to do it we can discuss our idea's Michael Devenijn DKMA bvba
RE: [Asterisk-Users] Cisco 7960
This one helped me a lot : http://www.loligo.com/asterisk/Cisco/79xx/ kind regards Michael Devenijn www.dkma.be Van: Micke Andersson [mailto:[EMAIL PROTECTED] Verzonden: vr 24/10/2003 9:33 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] Cisco 7960 I need some help with upgrading a 7960. Any of you guys familiar with that ? I friend of mine have a couple of 7960 , and would like to get 'em to work. /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
[Asterisk-Users] Tested 7905G
Justy to let you all know that i tested 7905G phone with a SIP image (latest download) and it works great ! for a reasonable price but with a good quality and a brand ... which inspires trust and helps selling better The only minus : Missing a microphone to work handsfree (or i didn't find it.) but strange enough their is a speaker ... Michael Devenijn IT DKMA
[Asterisk-Users] Cisco 7905G phones
I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael
